Hi,
here what i have:
[2001]--[Asterisk]---[ISDN-Trunk]---[PBX]--[8004]
Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 on
RedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL.
Dialing from Astersik extension 2001 to PBX extension 8004 via ISDN Trunk gives
me the following error,
-- Executing Dial("SIP/2001-8a8e",
"Modem/ttyI0:998004|20|r") in new stack
Oct 11 13:49:12 WARNING[262159]: channel.c:1901 ast_request: No channel type
registered for 'Modem'
Oct 11 13:49:12 NOTICE[262159]: app_dial.c:742 dial_exec: Unable to create
channel of type 'Modem'
== Everyone is busy/congested at this time
Oct 11 13:49:22 WARNING[262159]: pbx.c:1933 ast_pbx_run: Timeout, but no rule
't' in context 'default'
Extension 2001 gives "unreachable"
99XXXX is the code using for outgoing calls.
;sip.conf
[2001]
type=friend
secret=2001
auth=2001
callerid="user 2001" <2001>
host=dynamic
disallow=all
context=default
allow=ulaw
allow=alaw
;extensions.conf
[default]
exten => 2001,1,NoOp( call for ${EXTEN})
exten => 2001,2,Dial(SIP/${EXTEN},60,tr)
exten => 2001,3,Congestion
exten => _99.,1,Dial(Modem/ttyI0:${EXTEN:0},20,r)
;modem.conf
[interfaces]
context=remote
driver=i4l
language=en
type=autodetect
dialtype=tone
mode=ring
device => /dev/ttyI0
Have I missed something in my extensions.conf? or in modem.conf?
thanks for your support...
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HI! I configured asterisk to send all outgoing calls to our Gateway. I noticed when asterisk sends call to gateway that he represents all calls as "asterisk" and not as callerID(number of sjphone client registerd to asterisk). Can anyone give me an example of such configuration? Thank you
I am trying to route my calls through an outside IAX provider. I am having a problem with which codec to use. The only way I have successfully been able to make an outgoing call is if i do: disallow=all allow=g729 in the sip.conf file (for my phones) and the iax.conf file. The second I add one more codec to that list, for instance: disallow=all allow=g729 allow=ulaw I get the following error in the CLI: Nov 23 10:56:35 NOTICE[3799]: channel.c:1703 ast_set_write_format: Unable to find a path from ulaw to g729 Nov 23 10:56:35 NOTICE[3799]: channel.c:1736 ast_set_read_format: Unable to find a path from g729 to ulaw During this time, the number I am calling rings, however, when I pick up, the server hangs up and says this: -- IAX2/plainvoip/3 is ringing -- IAX2/plainvoip/3 stopped sounds -- IAX2/plainvoip/3 answered SIP/4035-0e93 Nov 23 10:56:41 WARNING[3799]: channel.c:2127 ast_channel_make_compatible: No path to translate from SIP/4035-0e93(4) to IAX2/plainvoip/3(256) Nov 23 10:56:41 WARNING[3799]: app_dial.c:1024 dial_exec: Had to drop call because I couldn't make SIP/4035-0e93 compatible with IAX2/plainvoip/3 -- Hungup 'IAX2/plainvoip/3' == Spawn extension (from-sip, 13102801234, 1) exited non-zero on 'sip/4035-0e93' I will post my configuration files if necessary. Thank you in advance for any help that anyone can offer. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051123/372e47dd/attachment.htm
On Nov 23, 2005, at 11:14 AM, Michael wrote:> I am trying to route my calls through an outside IAX provider.? I am > having a problem with which codec to use.? The only way I have > successfully been able to make an outgoing call is if i do: > > ? disallow=all > ? allow=g729 > > in the sip.conf file (for my phones) and the iax.conf file.? The > second I add one more codec to that list, for instance: > > ? disallow=all > ? allow=g729 > ? allow=ulaw > > I get the following error in the CLI: > > ? Nov 23 10:56:35 NOTICE[3799]: channel.c:1703 ast_set_write_format: > Unable to find a path from ulaw to g729 > ? Nov 23 10:56:35 NOTICE[3799]: channel.c:1736 ast_set_read_format: > Unable to find a path from g729 to ulaw ><snip> Asterisk cannot translate from other codecs to g729 UNLESS you buy a license for that technology. It can pass g729 along though. it looks like your service provider "plainvoice" only supports G729? I am a newb though so take this with a BIG grain of salt. Marty
Hi list, I've been trying all kinds of things for hours but I keep ending up with nothing, so I was hoping to get some help. Because I could not get it to work i'v completely reset to the default configuration, except for sip.conf If I call my number I get the DEMO talking to me so I know this works.. The problem is calling out. I want to drop a call file into the spool and have the server call me and if I answer connect me to the demo (if i can get that working i probably will be able to do the rest) Can anyone tell me what i'm doing wrong, what am I missing. Regards, Marius sip.conf /################################################### [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all ; First disallow all codecs allow=ilbc allow=g729 allow=gsm allow=ulaw allow=alaw allow=all ; Allow codecs in order of preference register => 31137110377:secret@budgetphone.nl/1000 [31137110377] type=friend context=default host=sip.budgetphone.nl fromuser=31137110377 fromdomain=sip.budgetphone.nl username=31137110377 insecure=very secret=secret qualify=no port=5060 ###################################################/ This is the call-file i'm dropping: /################################################### Channel: SIP/0031611111111@31137110377 Callerid: 31137110377 MaxRetries: 5 RetryTime: 300 WaitTime: 45 Context: default Extension: s Priority: 1 ################################################### / logfiles: /==> /var/log/asterisk/full <=Jun 20 15:28:16 VERBOSE[26387]: -- Attempting call on SIP/0031611111111@31137110377 for s@default:1 (Retry 1) Jun 20 15:28:16 DEBUG[26387]: Setting NAT on RTP to 0 Jun 20 15:28:16 DEBUG[26387]: Outgoing Call for 0031611111111 Jun 20 15:28:16 DEBUG[26387]: 0031611111111 is not a local user Jun 20 15:28:16 DEBUG[26387]: Acked pending invite 102 Jun 20 15:28:16 DEBUG[26387]: Stopping retransmission on '641f41e2315f7d5f00dd05c51b4bcabb@sip.budgetphone.nl' of Request 102: Found Jun 20 15:28:16 WARNING[26387]: Forbidden - wrong password on authentication for INVITE to '"31137110377" <sip:31137110377@sip.budgetphone.nl>;tag=as24baf051' Jun 20 15:28:16 DEBUG[26387]: update_user_counter(0031611111111) - decrement outUse counter Jun 20 15:28:16 DEBUG[26387]: 0031611111111 is not a local user Jun 20 15:28:16 NOTICE[26387]: Call failed to go through, reason 8 Jun 20 15:28:16 DEBUG[26387]: Stopping retransmission on '641f41e2315f7d5f00dd05c51b4bcabb@sip.budgetphone.nl' of Request 102: Found Jun 20 15:28:16 WARNING[26387]: Forbidden - wrong password on authentication for CANCEL ==> /var/log/asterisk/messages <=Jun 20 15:28:16 WARNING[26387]: Forbidden - wrong password on authentication for INVITE to '"31137110377" <sip:31137110377@sip.budgetphone.nl>;tag=as24baf051' Jun 20 15:28:16 NOTICE[26387]: Call failed to go through, reason 8 Jun 20 15:28:16 WARNING[26387]: Forbidden - wrong password on authentication for CANCEL ==> /var/log/asterisk/full <=Jun 20 15:28:31 DEBUG[26387]: Auto destroying call '641f41e2315f7d5f00dd05c51b4bcabb@sip.budgetphone.nl'/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060620/b117445e/attachment.htm